asterisk users - Oct 2020 - Multiple IP addresses and using same IP for outbound calls as inbound

Hello,

We have an Asterisk server with two public IP addresses, let's say 1.1.1.1
and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
dialled from Asterisk to an external destination. The external destination
sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP
is 1.1.1.1, which is great.

However if we receive a call in to 2.2.2.2 then the call dialled from
Asterisk to an external destination still comes from 1.1.1.1, whereas we
want it to come from 2.2.2.2. The source of any dialled call (the IP packet
and the SDP media address) should be the same as the address the related
inbound call was received to.

For example:
INVITE received to 1.1.1.1:5060 -> Asterisk dials
destination at termination.com -> INVITE sent from 1.1.1.1:5060 to
termination.com
INVITE received to 2.2.2.2:5060 -> Asterisk dials destination at pstn.com
->
INVITE sent from 2.2.2.2:5060 to pstn.com

Does anyone know how this can be achieved?

Thanks in advance for your help,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <dcunningham at
voisonics.com>
wrote:
> Hello,
>
> We have an Asterisk server with two public IP addresses, let's say
1.1.1.1
> and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
> dialled from Asterisk to an external destination. The external destination
> sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP
> is 1.1.1.1, which is great.
>
> However if we receive a call in to 2.2.2.2 then the call dialled from
> Asterisk to an external destination still comes from 1.1.1.1, whereas we
> want it to come from 2.2.2.2. The source of any dialled call (the IP packet
> and the SDP media address) should be the same as the address the related
> inbound call was received to.
>
> For example:
> INVITE received to 1.1.1.1:5060 -> Asterisk dials
> destination at termination.com -> INVITE sent from 1.1.1.1:5060 to
> termination.com
> INVITE received to 2.2.2.2:5060 -> Asterisk dials destination at
pstn.com ->
> INVITE sent from 2.2.2.2:5060 to pstn.com
>
> Does anyone know how this can be achieved?
>
If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names aren't
important as long as you can tell the difference.  Then explicitly
configure endpoint termination.com's "transport" parameter to
"transport-1.1.1.1" and pstn.com's "transport" parameter
to
"transport-2.2.2.2".   In your dialplan, you can see which endpoint
the
call came in on, and route it out the same endpoint.

If both providers are available from both interfaces, you can create 2
endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
same transports as above.




>
> Thanks in advance for your help,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
George Joseph
Asterisk Software Developer
direct/fax +1 256 428 6012
Check us out at www.sangoma.com and www.asterisk.org
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Hi George,

Thank you for the response. I'm a little unclear on what you mean by a
transport. We're using chan_sip, not pjsip.

Do you mean a device in sip.conf, using bindaddr to set the address to bind
for that device? We've only used bindaddr in the [general] section before,
but if it will work in a device that could be the answer.


On Fri, 23 Oct 2020 at 00:13, George Joseph <gjoseph at digium.com> wrote:
>
>
> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
> dcunningham at voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk server with two public IP addresses, let's say
>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged
with
>> a call dialled from Asterisk to an external destination. The external
>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>> address in the SDP is 1.1.1.1, which is great.
>>
>> However if we receive a call in to 2.2.2.2 then the call dialled from
>> Asterisk to an external destination still comes from 1.1.1.1, whereas
we
>> want it to come from 2.2.2.2. The source of any dialled call (the IP
packet
>> and the SDP media address) should be the same as the address the
related
>> inbound call was received to.
>>
>> For example:
>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>> destination at termination.com -> INVITE sent from 1.1.1.1:5060 to
>> termination.com
>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destination at
pstn.com
>> -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>
>> Does anyone know how this can be achieved?
>>
>
> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
> for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names
> aren't important as long as you can tell the difference.  Then
explicitly
> configure endpoint termination.com's "transport" parameter to
> "transport-1.1.1.1" and pstn.com's "transport"
parameter to
> "transport-2.2.2.2".   In your dialplan, you can see which
endpoint the
> call came in on, and route it out the same endpoint.
>
> If both providers are available from both interfaces, you can create 2
> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
> same transports as above.
>
>
>
>
>
>>
>> Thanks in advance for your help,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> George Joseph
> Asterisk Software Developer
> direct/fax +1 256 428 6012
> Check us out at www.sangoma.com and www.asterisk.org
> [image: image.png]
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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